1. Field of the Invention
The present invention relates to an audio digital-to-analog converting system, and more particularly, to an audio digital-to-analog converting system for lowering operations of sampling rate conversions by using a low-pass filter and method thereof.
2. Description of the Prior Art
In the architecture of an audio DAC system adopted in the industry presently, only processes of interpolation and filtering are applied to the inputted audio signals before the input terminal of a digital-to-analog converter. An audio playback function that supports different sampling rates is lacked.
In order to support the audio playback function with different sampling rates, the audio digital-to-analog converting system must be collocated with an accurate operating clock, wherein the operating clock must be integer-times of its sampling rate. Please refer to FIG. 1. FIG. 1 is a table 100 illustrating corresponding values between audio sampling rates and operating clocks of an audio digital-to-analog converting system. Familiar operating clocks include 256×Fs/384×Fs/512×Fs/768×Fs, wherein Fs represents the abovementioned sampling rate. However, within a general multimedia player, the operating clocks listed in the table 100 do not exist in the original system. To support the audio playback function with different sampling rates, a dedicated phase lock loop (PLL) circuit corresponding to the generated operating clocks must be added into this design, which will increase the manufacturing cost in hardware. Additionally, it is a difficult issue to handle the audio/video synchronization with a multimedia player if we adapt a specific clock rate with different audio sampling rates. With this invention, it provides a solution of the audio/video synchronization in the audio digital-to-analog system under one fixed clock rate following the video processor.
Plenty of methods for providing the function of converting the sampling rate to the audio digital-to-analog converting system under a fixed operating clock have already been disclosed by some scholars. Please refer to FIG. 2 to FIG. 4. FIG. 2, FIG. 3, and FIG. 4 are respectively a diagram showing a conventional digital-to-analog converting system with sampling rate conversions according to the prior art. In FIG. 2, the method of work is firstly converting an input signal In of the digital-to-analog converting system 200 into an audio signal with 48 KHz, wherein the filters H1(z) and H2(z) substantially increase the burden of the operations. Take converting 44.1 KHz into 48 KHz as an example, the stop-band gain is set as (−90 dB) under a condition that B=160 and A=147, and thus H1(z) is a 9665-order FIR filter. Even though the interpolation/decimation mechanism is divided into three segments (8-7, 10-7, 2-3) for processing, its cost for operations is still too high.
In FIG. 3, the method of work is using A, B, and H3(z) of the digital-to-analog converting system 300 to perform an interpolation on input signals with different sampling rates, wherein the filter H3(z) also increases the cost for operations. Take converting 44.1 KHz into 48 KHz as an example, the stop-band gain is set as (−90 dB) under a condition that B=2500 and A=147, and thus H3(z) is a 7917-order FIR filter. Even though the interpolation/decimation mechanism is divided into three segments (25-7, 10-7, 10-3) for processing, its cost for operations is still too high. If the filter H3(z) is directly removed and only the interpolation/decimation mechanism is performed, a large number of image signals may fall into the audio frequency range (20 Hz˜20 KHz) to cause the audio quality poor.
In FIG. 4, the digital-to-analog converting system 350 has been disclosed in the U.S. Pat. No. 6,834,292. Take converting 44.1 KHz into 48 KHz at the input end of the digital-to-analog converting system 350 as an example, parameters are set as below: R1=2, R2=32, R3=160, and S1=4704. Therefore, H11(z) is a 126-order FIR filter, H22(z) is a 143-order FIR filter, and H33(z) is a 3-order comb filter. Without considering the operations of H44(z), totally 193 multiplications are needed to complete one time of sampling frequency conversion. If the multiplications are replaced by the additions by using a Radix-4 Booth Multiplier, totally 1737 additions are needed to complete one time of sampling rate conversion.
As can be seen, numerous sampling rate converting mechanisms applied to the audio digital-to-analog converting system have been disclosed in the prior art, but their operations are too huge and cannot be omitted. In addition, their considerations on cost and the whole efficiency are not good enough. Hence, the audio digital-to-analog converting system presently needs to be improved to provide the function of converting sampling rates without increasing any dedicated PLL circuit.